search for: bockman

...risk, using pcm, au, or ul, I get pops in the audio. What am I doing wrong? Also, what types of wav (not WAV) files does Asterisk generate? I have a couple wav files to convert too. I believe if I do a sox file.wav -r 8000 -c 1 fle.wav file.gsm, I have the same type of problem. Thanks, Kevin Bockman

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there. Thanks, Kevin _____________________________________________________________ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com

...c on hold, anything extra. For testing, all I'm doing is making a call, playing a sound file, and doing a one-line database update about 3 times during the call. I have converted the sound file to ulaw so it does not transcode. If anyone has some hints, I would appreciate it. Thanks, Kevin Bockman

...Also, I'm not seeing any activity anymore on eth1. I used to, but even then it still didn't work. Do I not have the correct modules loaded? I'm pretty sure the how-to didn't go into deal about which should be loaded, just to "load the correct ones". Thanks, Kevin Bockman

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Hi. My bad, but I have not been tracking the -stable list and I just updated -stable on a production machine. It seemed to work just wonderful, as normal, for about 10 minutes then it started to deny connections and when you run commands, it would just hang. I have done a reboot and fsck everything. It is up now but things are still strange. Here's the output. I just updated -stable at

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

Sorry, this is a re-post. I sent the message from the wrong e-mail address. I was checking up on the cvs changes. Hi. I just tried setting up the send url function of Queue app, but it did not seem to work with gnophone. I must have missed something. It never tries to go to the page. I have this in extensions.conf: exten => 1,3,Queue(support,t,http://www.yahoo.com) It shows in the

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. >> >> So, I ran eagi-sphinx-test unde...

Guys'n'Gals vote for bug 3836 - Chanspy is back. Better than ever. Let's get this one into CVS. Julian

...39;ve not been able to get that to work - crashes my * box with CVS HEAD. > > -josiah It does this to me too. We should send in a bug report. I think there have been other people reporting it also. It would be "reeel nice-like" [Beverly Hillbillies] to have this working. Kevin Bockman

...to get meetme application registered? Do I have to recompile asterisk again ? I don't see the compiled meetme.so module in the directory. Regards, Sam -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin Bockman Sent: Friday, February 10, 2006 6:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How come I don't have the MeetMe applicationregistered? Anthony Azzopardi wrote: > How come I don't have the MeetMe application registered? You need a timin...

Good eye! Its getting late maybe I should just stop now Thank again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin Bockman Sent: Monday, March 13, 2006 8:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Unknown signalling method 'pri_cpe' Hall, Eric M. wrote: > [chan_zap.so] => (Zapata Telephony) > Mar 13 20:44:26 ERROR[10829]: chan_zap.c:10598 set...

Hi, Would anyone be kind enough to send me the 2.0.2 SIP firmware? I asked VoipSupply for it on Wednesday, nagged them again on Thursday and they did not even send the request yet. I was supposed to have it 'Friday morning' at the latest. I'm doing equipment upgrades this weekend so this is the time to do it. I've been having random phone crashing using 2.0.1. I also

...I will be continuing to muck with the source files, but I don't see what the problem is from here since linux/rtc.h should be included since I am running 2.6 and defined USE_RTC. I checked and /usr/include/linux/rtc.h is there and is the same as the one from 2.6.11.11 sources. Thanks, Kevin Bockman

...onger produces Wildcards. I'm not sure > why they don't work with our 6850 and the techs at Dell didn't know > either. Maybe they are not 100% PCI compliant. > > Matthew Roth > InterMedia Marketing Solutions > Software Engineer and Systems Developer > > Kevin Bockman wrote: > >> Chuck Bunn wrote: >> >>> Does anyone know if the Digium Wildcard will work on a PCI Express >>> or PCI-X motherboard. Specifically I am looking at the Dell 850 1U >>> rack server for use with Asterisk. >> >> >> >> They...

Hi. I have been experencing some filesystem problems for the last month or so. I was running 4.8-STABLE and updated to 5.1-RELEASE-p2. While I was running 4.8 and I tried to run a command that required hard disk activity, the process would 'hang' and I would no longer be able to ssh or telnet in. I would get stuck after typing in my login. Running 5.1 is a different story. I did a low